Bandwidth division encoding is a common method of encoding an audio signal at a low bit rate while still achieving a high quality playback signal. This is done by splitting an input audio signal into signals for plural frequency bands (subbands) using a band division filter, or by converting the input signal to a frequency domain signal using a Fourier transform or other time-frequency conversion algorithm, then dividing the signal into multiple subbands in the frequency domain, and allocating an appropriate coding bit to each of the bandwidth divisions. The reason why a high quality playback signal can be obtained from low bit rate data using bandwidth division encoding is that during the encoding process the signal is processed based on human acoustic sense characteristics.
Human auditory sensitivity at a frequency of approximately 10 kHz or greater generally drops, and low sound levels become difficult to hear. Furthermore, a phenomenon called “frequency masking” is well known. Due to frequency masking, when there is a high level sound in a particular frequency band, low level sounds in neighboring frequency bands become difficult to be audible. Allocating bits and encoding signals that are difficult to be sensed due to such auditory characteristics has substantially no effect on the quality of the playback signal, and therefore encoding such signals is meaningless. Conversely, by taking the code bits allocated to this audibly meaningless band and reallocating the bits to audibly sensitive subbands, audibly sensitive signals can be encoded with great detail, thereby effectively improving the quality of the playback signal.
An example of such coding using band division is MPEG-4 MC (ISO/IEC 14496-3) by international standard, which enables high quality coding of a 16 kHz or greater wideband stereo signal at an approximately 96 Kbps bit rate.
If the bit rate is lowered to, for example, approximately 48 Kbps, only a 10 kHz or shorter bandwidth can be encoded with high quality, resulting in muffled sound. One method of compensating for degraded sound quality resulting from such bandwidth limiting is called SBR (spectral band replication) and is described in the Digital Radio Mondiale (DRM) System Specification (ETSI TS 101 980) published by the European Telecommunication Standards Institute (ETSI). Similar technology is also disclosed, for example, in AES (Audio Engineering Society) convention papers 5553, 5559, 5560 (112th Convention, 2002 May 10–13, Munich, Germany).
SBR seeks to compensate for the high frequency band signals (referred to as high frequency components) that are lost by the audio encoding process such as MC or equivalent band limiting process. Signals in frequency bands below the SBR-compensated band (also called low frequency components) must be transmitted by some other means. Information for generating a pseudo-high frequency component based on the low frequency components transmitted by other means is contained in the SBR-coded data, and audio degradation due to band limiting can be compensated for by adding this pseudo-high frequency component to the low frequency components.
FIG. 7 is a schematic diagram of a decoder for SBR band expansion according to the prior art. Input bitstream 106 is separated into low frequency component information 107, high frequency component information 108, and added information 109. The low frequency component information 107 is, for example, information encoded using the MPEG-4 AAC or other coding method, and is decoded to generate a time signal representing the low frequency component. This time signal representing the low frequency component is divided into multiple subbands by analysis filter bank 103.
The analysis filter bank 103 is generally a filter bank that uses complex-valued coefficients, and the divided subband signal is represented as a complex-valued signal. Band expander 104 compensates for the high frequency component lost due to bandwidth limiting by copying low frequency subband signals representing low frequency components to high frequency subbands. The high frequency component information 108 input to the band expander 104 contains gain information for the compensated high frequency subband so that gain is adjusted for each generated high frequency subband.
The high frequency subband signal generated by the band expander 104 is then input with the low frequency subband signal to the synthesis filter bank 105 for band synthesis, and output signal 110 is generated. Because the subband signals input to the synthesis filter bank 105 are generally complex-valued signals, a complex-valued coefficient filter bank is used as the synthesis filter bank 105.